Adding BitrateController to audio network adaptor.

webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc

+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+
+#include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"

[hlundin-webrtc, 2016/09/15 11:01:09]

Goes before any other #includes.

[minyue-webrtc, 2016/09/15 12:12:15]

oh, really! I have been putting <xxx> at the top. will pay attention in the
future.

[hlundin-webrtc, 2016/09/19 11:08:57]

Yes. Include foo.h first of all in foo.cc.
https://google.github.io/styleguide/cppguide.html#Names_and_Order_of_Includes

+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+namespace {
+// L2(14B) + IPv4(20B) + UDP(8B) + RTP(12B) + SRTP_AUTH(10B) = 64B = 512 bits

[hlundin-webrtc, 2016/09/15 11:01:09]

This is dependent on transport type. Can it be made settable from higher layers?

[minyue-webrtc, 2016/09/15 12:12:15]

I'd prefer to keep it local since there is an effort in moving it to BWE, then
bitrate controller won't matter, so no API need to add.

[hlundin-webrtc, 2016/09/19 11:08:57]

Acknowledged.

[minyue-webrtc, 2016/09/19 13:16:30]

I discussed this with Michael again and realized that even with the planned work
in BWE, it would still be good for BitrateController to know and header size,
and the header size is better passed down from a higher level, preferably
through Controller::SetConstraints(). I added it a TODO here in the last patch
set, so as to keep this CL clean.

+constexpr int kPacketOverheadBits = 512;
+}
+
+BitrateController::Config::Config(int initial_bitrate_bps,
+                                  int initial_frame_length_ms)
+    : initial_bitrate_bps(initial_bitrate_bps),
+      initial_frame_length_ms(initial_frame_length_ms) {}
+
+BitrateController::Config::~Config() = default;
+
+BitrateController::BitrateController(const Config& config)
+    : config_(config),
+      bitrate_bps_(config_.initial_bitrate_bps),
+      overhead_rate_bps_(kPacketOverheadBits * 1000 /
+                         config_.initial_frame_length_ms) {
+  RTC_DCHECK_GT(bitrate_bps_, 0);
+  RTC_DCHECK_GT(overhead_rate_bps_, 0);
+}
+
+void BitrateController::MakeDecision(
+    const NetworkMetrics& metrics,
+    AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
+  // Decision on |bitrate_bps| should not have been made.
+  RTC_DCHECK(!config->bitrate_bps);
+
+  if (metrics.target_audio_bitrate_bps) {
+    int overhead_rate =
+        config->frame_length_ms
+            ? kPacketOverheadBits * 1000 / *config->frame_length_ms
+            : overhead_rate_bps_;
+    // If |metrics.target_audio_bitrate_bps| had included overhead, we would
+    // simply do:
+    //     bitrate_bps_ = metrics.target_audio_bitrate_bps - overhead_rate;
+    // Follow https://bugs.chromium.org/p/webrtc/issues/detail?id=6315 to track
+    // progress regarding this.
+    // Now we assume that |metrics.target_audio_bitrate_bps| can handle the
+    // overhead of most recent packets.
+    bitrate_bps_ = std::max(0, *metrics.target_audio_bitrate_bps +
+                                   overhead_rate_bps_ - overhead_rate);
+    // TODO(minyue): apply a smoothing on the |overhead_rate_bps_|.
+    overhead_rate_bps_ = overhead_rate;
+  }
+  config->bitrate_bps = rtc::Optional<int>(bitrate_bps_);
+}
+
+}  // namespace webrtc